Archives : March-2017
How to upgrade Asterisk from 10.x to whatever the recommended version is that will allow me to continue to use Fax For..
Hi!Im having a setup where my asterisk PBX connects to PSTN via a single SIP trunk.Now, when I transfer or redirect incoming calls from the SIPtrunk to another number which is routed through the SIP trunk, my asterisk stays on the way; it just di..
Ive upgraded to to asterisk 11.25.1 (from 1.8).My local asterisk is showing it is registered with remote asterisk (same version), But when I try to make a call I get:iax2 show registry HostdnsmgrUsernamePerceived RefreshState192.168.142.1:4569N home_serve192.168.142.7:456960Register..
I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP addresses from the same subnet set on one interface, and bindaddr is set to the second on them in sip.conf and in iax.conf. Incoming connections work as expected. However, for outgo..
I have been working on a project with asterisk and Kamailio. I would prefer using Kamailio because i have personally met with the developers and it has more active users and rapid developments. The developers are also very friendly and helpful. And w..
Want to integrate my cellular service into my asterisk dial plan, so this requires SIP capabilities such as one can get with Vitelity vMobile.Does Google Project Fi or Ting or another offer a SIP integration option, but not advertise it?A drawb..
Hellousing Asterisk 1.8.32.3Current music on hold :myserver*CLI> moh show files Class: default File: /var/lib/asterisk/moh/macroform-robot_dity File: /var/lib/asterisk/moh/macroform-cold_day File: /var/lib/asterisk/moh/reno_project-system File: /var/lib/asterisk/moh/manolo_camp-morning_cof..
Apologies if this is considered off-topic; I suspect the information might benefit a portion of the list.Can anyone point me in a direction to start implementation of E-911 services?Is this just something my upstream should supply, or can I connect..
I need to raise my ptime to 60 on my codecs for outbound calls. To that effect, I add on the endpoint disallow=all allow=ulaw:60and also use_avpf : false use_ptime: trueBut the invites always leave with ptime:20. It used to work fine in the old SIP chann..